Distributed processing networks are being increasingly used for live voice communications between network nodes using Voice over IP or VoIP technology. In VOIP technology, after the speech is digitized, the digitized speech is divided into packets, each packet including a header, and a data payload of one to several frames of encoded speech. Distributed processing networks for delivering the packets to desired endpoints are typically designed to provide a Best Effort or BE single service model that does not discriminate in packet delivery between services and does not control service access or quality. Quality of Service or QoS architectures have been developed for BE environments to provide guaranteed transmission characteristics end-to-end such as available bandwidth, maximum end-to-end delay, maximum end-to-end delay variation (jitter), and packet/cell loss levels to provide continuous data streams suitable for real-time phone calls and video conferencing. Such QoS architectures include protocols such as the Resource ReSerVation Protocol or RSVP and the Real-Time Transfer Protocol or RTP.
RSVP is a signaling protocol that guarantees receivers a requested end-to-end QoS. RSVP serves as an internet signaling protocol through the transmission of QoS parameters. Under RSVP, an end point negotiates with the network to allocate or reserve protected resources for traffic that the end point will generate or receive. The two messages that perform the reservation request and installation are the Path and Resv messages. Robustness is achieved through maintaining a soft state network by transmitting periodic refresh messages to maintain a reservation and path state along the reservation path. If the intermediate nodes do not receive the refresh message, the reservation will time out and be deleted.
RTP is a voice bearer channel transfer protocol. RTP neither guarantees a QoS nor provides for resource reservations. RTP runs on the transport layer of the Open Systems Interconnection or OSI model and defines a session by two components, namely its profile and payload format where the payload is the data being transmitted. The payload format specifies the format of the data within the RTP packet such as encoding and compression schemes. RTP functions include loss detection for quality estimation and rate adaptation, sequencing of data, intra- and intermedia synchronization, session identification using a session id, source identification using a synchronization source id or SSRC, and basic membership information.
The Real-Time Control Protocol or RTCP, a companion protocol to RTP, is used by applications to monitor the delivery of RTP streams. The joint operation of RTP and RTCP is illustrated by FIG. 1. Referring to FIG. 1, media packets transmitted between A 100 and B 104 and vice versa during a session are formatted and transmitted (continuously or frequently) over network 108 according to RTP while additional performance information governing the communication fink (e.g., key statistics about the media packets being sent and received by each end point (A or B) such as jitter, packet loss, round-trip time, etc.) are collected by the end points and transmitted (discontinuously or less frequently) over the network 108 to one another and to a session monitor 112 according to RTCP using IP multicast, unicast, or dual unicast techniques. End points A and B are typically computational components but can be or include any other form of audio or video communications interface. The network monitor can be, for example, VoIP Monitoring Manager™ or VMM™ by Avaya, Inc. RTCP performance information is useful not only for the session participants, A and B, but also for a network monitor 112. Network administrators can use such information not only for network administration but also for network troubleshooting and management.
Under either protocol, VoIP introduces a whole new range of QoS problems which were not previously significant or, in some cases, even encountered in circuit-switched networks. Voice telephony depends upon reliable, low latency, real-time delivery of audio data. In VoIP, values for latency, packet loss, and jitter can increase substantially, particularly during periods of heavy network traffic, causing a user to experience a much poorer quality of communication (e.g., audio or video distortion, unacceptable levels of asynchronization between audio and video streams, etc.) than would be experienced if the call were made by a traditional circuit-switched telephony network.
When a user experiences a poor quality of communication quality, there is no simple, user friendly method for him to log the details of the call (number dialed, date, and time), and such details are typically lost. IP hard- and soft-telephones calculate, during a session, the values of latency, packet loss, and jitter. Some telephones, such as the Cisco IP Phone 7960™, permit a user to press a button and view the current values for latency, packet loss, and jitter. However, such telephones fail to record the details of the call. In fact, much of the data collected from the end point(s) by the voice quality monitoring system, such as the session monitor in RTCP, is typically discarded with only a generalized summary of the session being retained for later use by network administration personnel. The loss of such data can hinder later diagnosis of the problem.
Current voice quality monitoring systems lack the capability of detemining when a problem is occurring in order to take a snapshot of the network. Taking a snapshot of the network attributes at the instant that the voice quality degraded will aid in instantaneous troubleshooting of the network. The user best knows when they are experiencing poor audio and/or video quality at a specific instant in time; however, when the network administration person later wants to troubleshoot by having the user replace the call, the user often fails to experience the same problem(s). To make matters worse, most users either fail to report problems due to the inconvenience of lodging a complaint and/or are unable to fully report the problem due to the inability of remembering who and when they were calling at the time they had the problem.